101 research outputs found

    Roaming Real-Time Applications - Mobility Services in IPv6 Networks

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    Emerging mobility standards within the next generation Internet Protocol, IPv6, promise to continuously operate devices roaming between IP networks. Associated with the paradigm of ubiquitous computing and communication, network technology is on the spot to deliver voice and videoconferencing as a standard internet solution. However, current roaming procedures are too slow, to remain seamless for real-time applications. Multicast mobility still waits for a convincing design. This paper investigates the temporal behaviour of mobile IPv6 with dedicated focus on topological impacts. Extending the hierarchical mobile IPv6 approach we suggest protocol improvements for a continuous handover, which may serve bidirectional multicast communication, as well. Along this line a multicast mobility concept is introduced as a service for clients and sources, as they are of dedicated importance in multipoint conferencing applications. The mechanisms introduced do not rely on assumptions of any specific multicast routing protocol in use.Comment: 15 pages, 5 figure

    Rapid Synchronisation of RTP Flows

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    A Novel Approach to Reduce the Unicast Bandwidth of an IPTV System in a High-Speed Access Network

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    Channel change time is a critical quality of experience (QOE) metric for IP-based video delivery systems such as Internet Protocol Television (IPTV). An interesting channel change acceleration scheme based on peer-assisted delivery was recently proposed, which consists of deploying one FCC server (Fast Channel Change Server) in the IP backbone in order to send the unicast stream to the STB (Set-Top Box) before sending the normal multicast stream after each channel change. However, deploying such a solution will cause high bandwidth usage in the network because of the huge unicast traffic sent by the FCC server to the STBs. In this paper, we propose a new solution to reduce the bandwidth occupancy of the unicast traffic, by deploying the FCC server capabilities on the user STB. This means that, after each channel change request, the STB will receive the unicast traffic from another STB instead of the central server. By using this method, the unicast traffic will not pass through the IP network; it will be a peer-to-peer communication via the Access Network only. Extensive simulation results are presented to demonstrate the robustness of our new solution

    Packet loss characteristics of IPTV-like traffic on residential links

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    Packet loss is one of the principal threats to quality of experience for IPTV systems. However, the packet loss characteristics of the residential access networks which carry IPTV are not widely understood. We present packet level measurements of streaming IPTV-like traffic over four residential access links, and describe the extent and nature of packet loss we encountered. We discuss the likely impact of these losses for IPTV traffic, and outline steps which can ameliorate this

    Service quality assurance for the IPTV networks

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    The objective of the proposed research is to design and evaluate end-to-end solutions to support the Quality of Experience (QoE) for the Internet Protocol Television (IPTV) service. IPTV is a system that integrates voice, video, and data delivery into a single Internet Protocol (IP) framework to enable interactive broadcasting services at the subscribers. It promises significant advantages for both service providers and subscribers. For instance, unlike conventional broadcasting systems, IPTV broadcasts will not be restricted by the limited number of channels in the broadcast/radio spectrum. Furthermore, IPTV will provide its subscribers with the opportunity to access and interact with a wide variety of high-quality on-demand video content over the Internet. However, these advantages come at the expense of stricter quality of service (QoS) requirements than traditional Internet applications. Since IPTV is considered as a real-time broadcast service over the Internet, the success of the IPTV service depends on the QoE perceived by the end-users. The characteristics of the video traffic as well as the high-quality requirements of the IPTV broadcast impose strict requirements on transmission delay. IPTV framework has to provide mechanisms to satisfy the stringent delay, jitter, and packet loss requirements of the IPTV service over lossy transmission channels with varying characteristics. The proposed research focuses on error recovery and channel change latency problems in IPTV networks. Our specific aim is to develop a content delivery framework that integrates content features, IPTV application requirements, and network characteristics in such a way that the network resource utilization can be optimized for the given constraints on the user perceived service quality. To achieve the desired QoE levels, the proposed research focuses on the design of resource optimal server-based and peer-assisted delivery techniques. First, by analyzing the tradeoffs on the use of proactive and reactive repair techniques, a solution that optimizes the error recovery overhead is proposed. Further analysis on the proposed solution is performed by also focusing on the use of multicast error recovery techniques. By investigating the tradeoffs on the use of network-assisted and client-based channel change solutions, distributed content delivery frameworks are proposed to optimize the error recovery performance. Next, bandwidth and latency tradeoffs associated with the use of concurrent delivery streams to support the IPTV channel change are analyzed, and the results are used to develop a resource-optimal channel change framework that greatly improves the latency performance in the network. For both problems studied in this research, scalability concerns for the IPTV service are addressed by properly integrating peer-based delivery techniques into server-based solutions.Ph.D

    Delivering video services over IP networks

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    The main goal pursued in this Thesis is to contribute towards the design and development of an end-to-end solution/system that would assist in reliable, consistence, less packet-loss delivery of high-quality video signals of pre-recorded presentations, training lectures, live events such as seminars over standard IP networks. This Thesis will focus on the existing Internet Service Provider, Oman Telecommunications Company (Omantel) and its best delivery of high-bandwidth data such as video to its Local and regional offices and departments over IP networks. This video-over-IP system aims to accumulate the technical scientific knowledge required to be able to offer high-quality video, which is fully scalable over IP networks. It aims to convert this knowledge into experimental prototypes, which, after the Thesis, can be developed into an integrated generic environment for Video-over-IP service development and content production. The objective is to initially define the functionality of content Services that can be incorporated into the operations of Oman telecommunications company networks. Then define the functional characteristics and system requirements necessary for the deployment of content streaming services over Omantel IP based networks. The design of this system would be combined with streaming high-quality video, while maintaining scalability and bandwidth efficiencies required for large-scale enterprise deployment. The design would encompass various components that are needed to capture, store and deliver streaming video to desktops. It will investigate on what is required to deliver quality video over Omantel IP networks and will recommend the actual products and solutions for achieving the end result

    A Semantic-Based Middleware for Multimedia Collaborative Applications

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    The Internet growth and the performance increase of desktop computers have enabled large-scale distributed multimedia applications. They are expected to grow in demand and services and their traffic volume will dominate. Real-time delivery, scalability, heterogeneity are some requirements of these applications that have motivated a revision of the traditional Internet services, the operating systems structures, and the software systems for supporting application development. This work proposes a Java-based lightweight middleware for the development of large-scale multimedia applications. The middleware offers four services for multimedia applications. First, it provides two scalable lightweight protocols for floor control. One follows a centralized model that easily integrates with centralized resources such as a shared too], and the other is a distributed protocol targeted to distributed resources such as audio. Scalability is achieved by periodically multicasting a heartbeat that conveys state information used by clients to request the resource via temporary TCP connections. Second, it supports intra- and inter-stream synchronization algorithms and policies. We introduce the concept of virtual observer, which perceives the session as being in the same room with a sender. We avoid the need for globally synchronized clocks by introducing the concept of user\u27s multimedia presence, which defines a new manner for combining streams coming from multiple sites. It includes a novel algorithm for estimation and removal of clock skew. In addition, it supports event-driven asynchronous message reception, quality of service measures, and traffic rate control. Finally, the middleware provides support for data sharing via a resilient and scalable protocol for transmission of images that can dynamically change in content and size. The effectiveness of the middleware components is shown with the implementation of Odust, a prototypical sharing tool application built on top of the middleware

    Distribuição de vídeo para grupos de utilizadores em redes móveis heterogéneas19

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    The evolutions veri ed in mobile devices capabilities (storage capacity, screen resolution, processor, etc.) over the last years led to a signi cant change in mobile user behavior, with the consumption and creation of multimedia content becoming more common, in particular video tra c. Consequently, mobile operator networks, despite being the target of architectural evolutions and improvements over several parameters (such as capacity, transmission and reception performance, amongst others), also increasingly become more frequently challenged by performance aspects associated to the nature of video tra c, whether by the demanding requirements associated to that service, or by its volume increase in such networks. This Thesis proposes modi cations to the mobile architecture towards a more e cient video broadcasting, de ning and developing mechanisms applicable to the network, or to the mobile terminal. Particularly, heterogeneous networks multicast IP mobility supported scenarios are focused, emphasizing their application over di erent access technologies. The suggested changes are applicable to mobile or static user scenarios, whether it performs the role of receiver or source of the video tra c. Similarly, the de ned mechanisms propose solutions targeting operators with di erent video broadcasting goals, or whose networks have di erent characteristics. The pursued methodology combined an experimental evaluation executed over physical testbeds, with the mathematical evaluation using network simulation, allowing the veri cation of its impact on the optimization of video reception in mobile terminalsA evolução veri cada nas características dos dispositivos moveis (capacidade de armazenamento, resolução do ecrã, processador, etc.) durante os últimos anos levou a uma alteração signi cativa nos comportamentos dos utilizadores, sendo agora comum o consumo e produção de conteúdos multimédia envolvendo terminais móveis, em particular o tráfego vídeo. Consequentemente, as redes de operador móvel, embora tendo também sido alvo constante de evoluções arquitecturais e melhorias em vários parâmetros (tais como capacidade, ritmo de transmissão/recepção, entre outros), vêemse cada vez mais frequentemente desa adas por aspectos de desempenho associados à natureza do tráfego de vídeo, seja pela exigência de requisitos associados a esse serviço, quer pelo aumento do volume do mesmo nesse tipo de redes. Esta Tese propôe alterações à arquitetura móvel para a disseminação de vídeo mais e ciente, de nindo e desenvolvendo mecanismos aplicáveis à rede, ou ao utilizador móvel. Em particular, são focados cenários suportados por IP multicast em redes móveis heterogéneas, isto é, com ênfase na aplicação destes mecanismos sobre diferentes tecnologias de acesso. As alterações sugeridas aplicam-se a cenários de utilizador estático ou móvel, sendo este a fonte ou receptor do tráfego vídeo. Da mesma forma, são propostas soluções tendo em vista operadores com diferentes objectivos de disseminação de vídeo, ou cujas redes têm diferentes características. A metodologia utilizada combinou a avaliação experimental em testbeds físicas com a avaliação matemática em simulações de redes, e permitiu veri car o impacto sobre a optimização da recepção de vídeo em terminais móveisPrograma Doutoral em Telecomunicaçõe

    Interoperability of wireless communication technologies in hybrid networks : evaluation of end-to-end interoperability issues and quality of service requirements

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    Hybrid Networks employing wireless communication technologies have nowadays brought closer the vision of communication “anywhere, any time with anyone”. Such communication technologies consist of various standards, protocols, architectures, characteristics, models, devices, modulation and coding techniques. All these different technologies naturally may share some common characteristics, but there are also many important differences. New advances in these technologies are emerging very rapidly, with the advent of new models, characteristics, protocols and architectures. This rapid evolution imposes many challenges and issues to be addressed, and of particular importance are the interoperability issues of the following wireless technologies: Wireless Fidelity (Wi-Fi) IEEE802.11, Worldwide Interoperability for Microwave Access (WiMAX) IEEE 802.16, Single Channel per Carrier (SCPC), Digital Video Broadcasting of Satellite (DVB-S/DVB-S2), and Digital Video Broadcasting Return Channel through Satellite (DVB-RCS). Due to the differences amongst wireless technologies, these technologies do not generally interoperate easily with each other because of various interoperability and Quality of Service (QoS) issues. The aim of this study is to assess and investigate end-to-end interoperability issues and QoS requirements, such as bandwidth, delays, jitter, latency, packet loss, throughput, TCP performance, UDP performance, unicast and multicast services and availability, on hybrid wireless communication networks (employing both satellite broadband and terrestrial wireless technologies). The thesis provides an introduction to wireless communication technologies followed by a review of previous research studies on Hybrid Networks (both satellite and terrestrial wireless technologies, particularly Wi-Fi, WiMAX, DVB-RCS, and SCPC). Previous studies have discussed Wi-Fi, WiMAX, DVB-RCS, SCPC and 3G technologies and their standards as well as their properties and characteristics, such as operating frequency, bandwidth, data rate, basic configuration, coverage, power, interference, social issues, security problems, physical and MAC layer design and development issues. Although some previous studies provide valuable contributions to this area of research, they are limited to link layer characteristics, TCP performance, delay, bandwidth, capacity, data rate, and throughput. None of the studies cover all aspects of end-to-end interoperability issues and QoS requirements; such as bandwidth, delay, jitter, latency, packet loss, link performance, TCP and UDP performance, unicast and multicast performance, at end-to-end level, on Hybrid wireless networks. Interoperability issues are discussed in detail and a comparison of the different technologies and protocols was done using appropriate testing tools, assessing various performance measures including: bandwidth, delay, jitter, latency, packet loss, throughput and availability testing. The standards, protocol suite/ models and architectures for Wi-Fi, WiMAX, DVB-RCS, SCPC, alongside with different platforms and applications, are discussed and compared. Using a robust approach, which includes a new testing methodology and a generic test plan, the testing was conducted using various realistic test scenarios on real networks, comprising variable numbers and types of nodes. The data, traces, packets, and files were captured from various live scenarios and sites. The test results were analysed in order to measure and compare the characteristics of wireless technologies, devices, protocols and applications. The motivation of this research is to study all the end-to-end interoperability issues and Quality of Service requirements for rapidly growing Hybrid Networks in a comprehensive and systematic way. The significance of this research is that it is based on a comprehensive and systematic investigation of issues and facts, instead of hypothetical ideas/scenarios or simulations, which informed the design of a test methodology for empirical data gathering by real network testing, suitable for the measurement of hybrid network single-link or end-to-end issues using proven test tools. This systematic investigation of the issues encompasses an extensive series of tests measuring delay, jitter, packet loss, bandwidth, throughput, availability, performance of audio and video session, multicast and unicast performance, and stress testing. This testing covers most common test scenarios in hybrid networks and gives recommendations in achieving good end-to-end interoperability and QoS in hybrid networks. Contributions of study include the identification of gaps in the research, a description of interoperability issues, a comparison of most common test tools, the development of a generic test plan, a new testing process and methodology, analysis and network design recommendations for end-to-end interoperability issues and QoS requirements. This covers the complete cycle of this research. It is found that UDP is more suitable for hybrid wireless network as compared to TCP, particularly for the demanding applications considered, since TCP presents significant problems for multimedia and live traffic which requires strict QoS requirements on delay, jitter, packet loss and bandwidth. The main bottleneck for satellite communication is the delay of approximately 600 to 680 ms due to the long distance factor (and the finite speed of light) when communicating over geostationary satellites. The delay and packet loss can be controlled using various methods, such as traffic classification, traffic prioritization, congestion control, buffer management, using delay compensator, protocol compensator, developing automatic request technique, flow scheduling, and bandwidth allocation.EThOS - Electronic Theses Online ServiceGBUnited Kingdo
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